SIP FAQ

SIP Basics and Pricing

What is SIP Trunking and what is it useful for?

SIP Trunking is one of the services we offer made to match heavy calling needs, both outbound or inbound. To get a better grasp of what SIP Trunking is and what its use, please read our dedicated article:SIP Trunking Use Case Presentation

While SIP is mostly used for call termination (outbound call), you can do both origination (inbound) and termination (outbound) with CALLR's SIP trunk. We offer many features for inbound calls handling using our API such as custom IVR diagram and call tracking.

We have a worldwide data center presence to ensure an unparalleled service quality and security for our clients.

Do I have to pay to set up my SIP Trunk with CALLR?

Basic installations are free of charge. Once your account is created, you just have to configure your PBX to use CALLR's services and begin testing right away. We ensure a free support for basic installations but installations requiring an extensive intervention of our support services will be charged.

What is the pricing model for SIP?

SIP trunks are free of charge: we do not charge setup or monthly fee. All our prices are displayed on the pricing section of our website. We offer a discount for high volumes customers, contact our sales services to know if you are eligible.

What are the supported PBX?

CALLR SIP can be used with any VoIP software. Here is an overview of our trunk compatibility:

Vendor

Type

Supported

FreeSwitch

Softswitch

Yes

Asterisk

IP-PBX

Yes

Free PBX

IP-PBX

Yes

3CX

IP-PBX

Yes

Elastix

IP-PBX

Yes

Telestax

IP-PBX

Yes

Vicidial

IP-PBX

Yes

Cisco

E-SBC

Yes

Avaya

E-SBC

Yes

Number and CLI Management

Can I buy numbers for my trunk?

You can buy DIDs at any time from a large selection of countries and number types. You can even automate the process for some countries and number types using the API. Read our dedicated article to know more about buying numbers at CALLR. If you are looking for number types or countries that are not available on our API, or short, special, premium or toll-free numbers feel free to contact the support team to ask them about it.Numbers at CALLR - Buying numbers (DIDs)

How should I format the number I dial?

The format of the numbers dialed in your VoIP installation must stay consistent. You can choose between three different formatting while setting up your SIP trunk:

Can I show a specific CLI while calling?

You can use any number you own as CLI. You can purchase and assign numbers dynamically using our API. For more information about numbers and CLIs handling, check our dedicated article: Numbers at CALLR.

Can I port numbers in?

You can port existing phone numbers to CALLR and use them for both outbound and inbound calls. The procedure depends on the phone numbers' country. If your need is not covered by the country detailed in the article below, please contact the support team to process your request. Learn more about the number portability process by reading our dedicated article:Number Portability How-to.

Are emergency numbers handled with CALLR SIP?

Emergency calls come with special requirements that are currently not addressed by our services. Therefore, emergency numbers are not handled by CALLR.

Internet requirements

How much bandwidth do I need?

Bandwidth needs depend on the codec you use. G711, a lossless codec, requires a lot of bandwidth: a standard ADSL connection will support up to 10-12 channels only. On the other hand, because G729 requires much less bandwidth, the same connection using this codec should support up to 100 channels. Please read our article on codecs to know more about them:Codecs Overview

Can I use a dynamic IP with SIP Trunk?

Trunk SIP uses authentification by IP, it won't work with a dynamic IP. Indeed, you need a static IP to be able to authenticate yourself while using trunk SIP. However, IAX uses the Password Authentication Protocol for authentification, so you can use IAX with a dynamic IP.

What are the issues between NAT and VoIP?

NAT consists of hiding (private) IP addresses behind another public IP address. To do so, all outgoing traffic is routed through a NAT router that replaces the source IP address with its own public address. NAT allows multiple devices on a LAN network to share a single public IP address.

SIP trunk uses two data streams for signaling and audio payloads: the signaling protocol is separated from the audio and worse, the port on which the audio traffic is sent is random. Therefore, even if the NAT router is able to handle the signaling traffic, it has no way of knowing it's paired with an audio stream that should be sent to the same device.

From an end-user perspective, it might translate into a partial (one-way audio) or complete absence of the audio signal on their calls despite a proper signaling: the telephone rings, the caller hears the ringing feedback, the called person sees the caller ID and ringing tone stops when the called person pick up the phone.

Because IAX carries both the signaling and the audio using only one stream, it has no issue with NAT.

VoIP-Info.org offers a comprehensive overview of the issues with NAT and VoIP as well an exhaustive list of solutions to either avoid the problem altogether (not use NAT) or work around the problem (properly setup NAT to work with VoIP).

Channels and gateways

How can I use the same gateway for multiple clients?

At CALLR, each SIP trunk is identified by its IP address. You can share the same gateway using an ''account code". The account code is an optional setup specified in the SIP header.

You can have multiple account codes for the same IP, allowing you to mutualize a gateway between different clients. Each client is identified by its account code and the gateway by the IP. Contact our support team to get started.

Can I receive incoming calls on several gateways?

Routing incoming calls to several gateways have two main interest: to have a failover gateway in case of failure of the main gateway or to balance calls load between two gateways.

Incoming calls are routed to a specific gateway. In the case of maintenance, you can route your incoming calls to a different gateway with a setting change made on the web interface, using the API or by contacting our support team.

Regarding failover gateways, a load balancing solution will soon be available through the API and the web interface. You can build an automated failover solution for yourself right now using our API and a simple script:

You'll need a script to check if the main gateway is available regularly. If not, you can change the gateway to the second gateway with a simple API command.

Can I manage my channels?

Yes, channels are free of charge and are attributed based on your calling needs. You can request additional channels by contacting the support team. To know more about channels, please read our dedicated article:Channel Management

Can I be a SIP reseller with CALLR?

You can indeed resell our SIP services. We created a specific web interface for SIP reseller where they can manage their customers. With this interface, you can create and manage SIP and IAX accounts for your customers and distribute your channels between your customers.